Method and system for optimizing the low-frequency sound rendition of an audio signal

ABSTRACT

A system and method for optimizing the low-frequency sound rendition of an audio signal, implementing variations in a plurality of parameters of the audio signal according to the volume level of the signal chosen by a user, in particular filtering or compression parameters, or parameters relating to the harmonics of the audio signal, while seeking to optimize the dynamics and the bandwidth of the audio signal according to the volume, in order to provide an optimal rendition to the user.

CROSS REFERENCE TO RELATED APPLICATIONS

This application is the National Stage of International Application No.PCT/FR2017/051599, having an International Filing date of 19 Jun. 2017,which designated the United States of America, and which InternationalApplication was published under PCT Article 21(2) as WO Publication No.2017/220906 A1, and which claims priority from, and the benefit of,French Application No. 1655698, filed on 20 Jun. 2016, the disclosuresof which are incorporated herein by reference in their entireties.

BACKGROUND 1. FIELD

The present disclosure relates to the field of processing sound signals.

The present disclosure more particularly relates to a method and asystem for optimising the low-frequency sound rendition of an audiosignal.

2. BRIEF DESCRIPTION OF RELATED DEVELOPMENTS

Methods and systems are known in the prior art for improving the soundrendition in the low frequency range.

The “low frequency range” is understood in the present disclosure to befrequencies of less than 150 Hz.

A method known in the prior art in the international patent applicationpublished under number WO 2013/068359 (ARKAMYS) relates to a method forreducing parasitic vibrations of a loudspeaker environment andassociated processing device. This invention of the prior artsubstantially concerns a method for reducing parasitic vibrations of aloudspeaker environment while maintaining the perception of the lowfrequencies of an electric sound signal, called the original soundsignal, intended to be broadcast after processing by said loudspeakerhaving a cut-off frequency, comprising the following steps: identifyinga frequency band that causes the loudspeaker to vibrate, called thevibration frequency band, isolating a low-frequency band of the originalsound signal having a frequency close to the cut-off frequency of theloudspeaker as the upper limit, generating at least one harmonic signalfrom the isolated low frequency band of the original sound signal,combining the original sound signal and the harmonic signal to obtain arecombined signal, removing the vibration frequency band from therecombined signal to obtain a signal that can be broadcast by theloudspeaker.

Methods are also known in the prior art for varying the gain in a staticmanner, in order to adjust the audio signal in the low frequency range.

SUMMARY

The purpose of the present disclosure is to overcome the drawbacks ofthe prior art by proposing a method and a system for optimising thelow-frequency sound rendition of an audio signal, by varying numerousparameters according to the audio volume.

For this purpose, the present disclosure relates, in the broadest sensethereof, to a method for optimising the low-frequency sound rendition ofan audio signal, implementing variations in a plurality of parameters ofsaid audio signal according to the volume level of said audio signalchosen by a user, in particular filtering or compression parameters, orparameters relating to the harmonics of said audio signal, while seekingto optimise the dynamic range and the bandwidth of said audio signalaccording to the volume, in order to provide an optimal rendition to theuser.

As opposed to the methods and systems known in the prior art, the methodaccording to the present disclosure varies numerous parameters of theaudio signal in order to provide an optimal rendition to the user.

Within the scope of the method according to the present disclosure, theaim is to optimise the dynamic range and the bandwidth of said audiosignal according to the volume, in order to provide an optimal renditionto the user, at both a low and high volume, without being detrimental tothe power handling of the system for reproducing said audio signal.

This plurality of parameters is adjusted, then stored by a personskilled in the art, then retrieved by the reproduction system as afunction of the volume level chosen by the user.

Thus, the dynamic range is increased at a low volume level, whileprotecting the reproduction system at a high volume level. This thusimproves the dynamic range of the low-frequency signal lost whenlistening at a low volume level as a result of the loudness effect ofthe human ear, while protecting the loudspeakers at a high volume level.This last point is an improvement on the aforementioned patent WO2013/06835 (ARKAMYS), in that the parasites can be reduced at a highvolume level, while optimising the performance of the loudspeakers at alow volume level.

According to one embodiment, said parameters are included in the groupconsisting of: the compression rate, the release time, the attack time,the recovery time, the threshold and the gain compensation.

Preferably, said method comprises a step of adjusting thecharacteristics of a high-pass filter according to the input level ofsaid audio signal.

The present disclosure further relates to a system for optimising thelow-frequency sound rendition of an audio signal, comprising means forvarying a plurality of parameters of said audio signal according to thevolume level of said audio signal chosen by a user, in particularfiltering or compression parameters, or parameters relating to theharmonics of said audio signal, while seeking to optimise the dynamicrange and the bandwidth of said audio signal according to the volume, inorder to provide an optimal rendition to the user.

Advantageously, said parameters are included in the group consisting of:the compression rate, the release time, the attack time, the recoverytime, the threshold and the gain compensation.

Preferably, said system comprises means for adjusting thecharacteristics of said high-pass filter according to the input level ofsaid audio signal.

BRIEF DESCRIPTION OF THE DRAWINGS

The present disclosure will be better understood after reading thedescription, provided for illustration purposes only, of one embodimentof the present disclosure, with reference to the Figures, in which:

FIG. 1 shows the system according to the present disclosure, in oneembodiment thereof.

DETAILED DESCRIPTION

The present disclosure relates to a method for optimising thelow-frequency sound rendition of an audio signal, implementingvariations in a plurality of parameters P1, P2, P3, . . . , PN of saidaudio signal according to the volume level V1, V2, . . . , VP of saidaudio signal chosen by a user, in particular filtering or compressionparameters, or parameters relating to the harmonics of said audiosignal, while seeking to optimise the dynamic range and the bandwidth ofsaid audio signal according to the volume, in order to provide anoptimal rendition to the user.

As opposed to the methods and systems known in the prior art, the methodaccording to the present disclosure varies numerous parameters of theaudio signal in order to provide an optimal rendition to the user.

Within the scope of the method according to the present disclosure, theaim is to optimise the dynamic range and the bandwidth of said audiosignal according to the volume, in order to provide an optimal renditionto the user, at both a low and high volume, without being detrimental tothe power handling of the system for reproducing said audio signal.

This plurality of parameters is adjusted, then stored by a personskilled in the art, then retrieved by the reproduction system as afunction of the volume level chosen by the user.

Thus, the dynamic range is increased at a low volume level, whileprotecting the reproduction system at a high volume level.

For example, for a volume variation ranging from 0 (minimum volume=−100dB of attenuation) to 30 (maximum volume=0 dB of attenuation), sevensteps are considered here for adjusting the parameters of the elementsin FIG. 1. The intermediate parameters are interpolated here.

With reference to FIG. 1, the following table is obtained:

TABLE 1 Volume level Parameters 0 5 10 15 20 25 30 14 14 × 0 14 × 5 14 ×10 14 × 15 14 × 20 14 × 25 14 × 30 40 40 × 0 40 × 5 40 × 10 40 × 15 40 ×20 40 × 25 40 × 30 41 41 × 0 41 × 5 41 × 10 41 × 15 41 × 20 41 × 25 41 ×30 42 42 × 0 42 × 5 42 × 10 42 × 15 42 × 20 42 × 25 42 × 30 16 16 × 0 16× 5 16 × 10 16 × 15 16 × 20 16 × 25 16 × 30 17 17 × 0 17 × 5 17 × 10 17× 15 17 × 20 17 × 25 17 × 30 51 51 × 0 51 × 5 51 × 10 51 × 15 51 × 20 51× 25 51 × 30 52 52 × 0 52 × 5 52 × 10 52 × 15 52 × 20 52 × 25 52 × 30 1919 × 0 19 × 5 19 × 10 19 × 15 19 × 20 19 × 25 19 × 30 61 61 × 0 61 × 561 × 10 61 × 15 61 × 20 61 × 25 61 × 30 62 62 × 0 62 × 5 62 × 10 62 × 1562 × 20 62 × 25 62 × 30 21 21 × 0 21 × 5 21 × 10 21 × 15 21 × 20 21 × 2521 × 30 22 22 × 0 22 × 5 22 × 10 22 × 15 22 × 20 22 × 25 22 × 30

It should be noted that the parameters mentioned in the left-hand columnof Table 1 above correspond to the blocks shown in FIG. 1.

If thirty adjustment steps are considered, a plurality of parameters(14×0, 14×1, 14×2, 14×3, etc.) thus corresponds to each volume level.

In one embodiment, said parameters P1, P2, P3, . . . , PN are includedin the group consisting of: the compression rate, the release time, theattack time, the recovery time, the threshold and the gain compensation.

In one embodiment, said method comprises a step of adjusting thecharacteristics of said high-pass filter F according to the input levelof said audio signal.

The present disclosure further relates to a system for optimising thelow-frequency sound rendition of an audio signal, comprising means forvarying a plurality of parameters P1, P2, P3, . . . , PN of said audiosignal according to the volume level V1, V2, . . . , VP of said audiosignal chosen by a user, in particular filtering or compressionparameters, or parameters relating to the harmonics of said audiosignal, while seeking to optimise the dynamic range and the bandwidth ofsaid audio signal according to the volume, in order to provide anoptimal rendition to the user.

In one embodiment, said parameters P1, P2, P3, . . . , PN are includedin the group consisting of: the compression rate, the release time, theattack time, the recovery time, the threshold and the gain compensation.

In one embodiment, said system comprises means for adjusting thecharacteristics of said high-pass filter F according to the input levelof said audio signal.

FIG. 1 shows the system according to the present disclosure, in oneembodiment thereof.

FIG. 1 shows that the system according to the present disclosurecomprises an input signal 10 and an output signal 30.

The input signal 10 is composed of a left channel 11 and a right channel12.

The output signal 30 is composed of a front left channel 31, a frontright channel 32, a back left channel 33 and a back right channel 34.

The system according to the present disclosure comprises a so-called“by-pass” element 13, a delay module 14 and a harmoniser 15.

The system according to the present disclosure comprises two gainmodules: a so-called “dry gain” module 16 and a so-called “wet gain”module 17.

The system according to the present disclosure further comprises aso-called “crossover” module 18, a second delay module 19 and acompression block 20.

Finally, said system according to the present disclosure comprises twofurther gain modules, a front low-frequency gain module 21, and a rearlow-frequency module 22, situated before the output.

The harmoniser 15 is composed of:

-   a pre-harmonic filter 40 of the type IIR (Infinite Impulse Response)    constituted in one example embodiment of two identical biquad    low-pass filters;-   a harmonic generator 41 (if the input signal into this module is    negative, the output signal of this module is zero); and-   a post-harmonic filter 42 of the type IIR (Infinite Impulse    Response) constituted in one example embodiment of one biquad    high-pass filter and of two identical biquad low-pass filters.    The so-called “crossover” module 18 is composed of:-   a so-called “crossover high” sub-module 51: a filter of the type IIR    (Infinite Impulse Response) constituted in one example embodiment of    two identical biquad high-pass filters; and-   a so-called “crossover low” sub-module 52: 51: a filter of the type    IIR (Infinite Impulse Response) constituted in one example    embodiment of two identical biquad low-pass filters.

Said compression block 20 is composed of:

-   a so-called “speaker cut” sub-module 61 constituted of a biquad    filter; and-   another sub-module 62, which is a stereo compressor, comprising    adjustable parameters such as the threshold, the compression slope,    the expansion slope, the attack time, the release time and the gain    compensation.

The above description of the present disclosure is provided for thepurposes of illustration only. It is understood that a person skilled inthe art can produce different variations of the present disclosurewithout leaving the scope of the patent.

What is claimed is:
 1. A method for optimising the low-frequency soundrendition of an audio signal, characterised in that it implementsvariations in a plurality of parameters of said audio signal accordingto the volume level of said audio signal chosen by a user, in particularfiltering or compression parameters, or parameters relating to theharmonics of said audio signal, while seeking to optimise the dynamicrange and the bandwidth of said audio signal according to the volume, inorder to provide an optimal rendition to the user.
 2. The method foroptimising the low-frequency sound rendition of an audio signalaccording to claim 1, characterised in that said parameters are includedin the group consisting of: the compression rate, the release time, theattack time, the recovery time, the threshold and the gain compensation.3. The method for optimising the low-frequency sound rendition of anaudio signal according to claim 1, characterised in that it comprises astep of adjusting the characteristics of a high-pass filter (F)according to the input level of said audio signal.
 4. A system foroptimising the low-frequency sound rendition of an audio signal,characterised in that it comprises means for varying a plurality ofparameters of said audio signal according to the volume level of saidaudio signal chosen by a user, in particular filtering or compressionparameters, or parameters relating to the harmonics of said audiosignal, while seeking to optimise the dynamic range and the bandwidth ofsaid audio signal according to the volume, in order to provide anoptimal rendition to the user.
 5. The system for optimising thelow-frequency sound rendition of an audio signal according to claim 4,characterised in that said parameters are included in the groupconsisting of: the compression rate, the release time, the attack time,the recovery time, the threshold and the gain compensation.
 6. Thesystem for optimising the low-frequency sound rendition of an audiosignal according to claim 4, characterised in that it comprises meansfor adjusting the characteristics of a high-pass filter according to theinput level of said audio signal.